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Francesco Peeters wrote:
<blockquote cite="mid:4A9EFF12.9090201@fampeeters.com" type="cite">
<pre wrap="">Francesco Peeters wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Does anybody else see the same behavior for VoipBuster connections?
When I trace one of the other SIP peers, I see it sends this message:
----------------------------------------------------------------------
<--- SIP read from 82.101.62.99:5060 --->
SIP/2.0 180 Ringing
Allow: INVITE,ACK,BYE,CANCEL,PRACK,SUBSCRIBE,NOTIFY,UPDATE
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:740540ee64fa957513ce89f03ef5e3f2@sip.xs4all.nl">740540ee64fa957513ce89f03ef5e3f2@sip.xs4all.nl</a>
Contact: <sip:82.101.62.99:5060>
Content-Type: application/sdp
CSeq: 103 INVITE
From: "**********" <a class="moz-txt-link-rfc2396E" href="mailto:sip:**********@sip.xs4all.nl"><sip:**********@sip.xs4all.nl></a>;tag=as70e84199
Record-Route:
<sip:82.101.62.115;lr;r2=on;ftag=as70e84199>,<sip:82.101.63.5;lr;r2=on;ftag=as70e84199>
Server: Cirpack/v4.41b (gw_sip)
To: <a class="moz-txt-link-rfc2396E" href="mailto:sip:0031********@sip.xs4all.nl"><sip:0031********@sip.xs4all.nl></a>;tag=00-08168-044b6f36-245cd72c7
Via: SIP/2.0/UDP
***.***.***.***:5060;received=***.***.***.***;rport=5060;branch=z9hG4bK07c2ed92
Content-Length: 182
v=0
o=cp10 125193221174 125193221174 IN IP4 82.101.62.66
s=SIP Call
c=IN IP4 194.109.8.2
t=0 0
m=audio 36984 RTP/AVP 8
b=AS:64
a=rtpmap:8 PCMA/8000/1
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 10 lines) ---
Found RTP audio format 8
Peer audio RTP is at port 194.109.8.2:36984
Found audio description format PCMA for ID 8
Capabilities: us - 0x10a (gsm|alaw|g729), peer - audio=0x8
(alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing),
combined - 0x0 (nothing)
Peer audio RTP is at port 194.109.8.2:36984
-- SIP/*********-089ca9b8 is ringing
-- SIP/*********-089ca9b8 is making progress passing it to
IAX2/2104-2287
Scheduling destruction of SIP dialog
'<a class="moz-txt-link-abbreviated" href="mailto:740540ee64fa957513ce89f03ef5e3f2@sip.xs4all.nl">740540ee64fa957513ce89f03ef5e3f2@sip.xs4all.nl</a>' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 82.101.62.99:5060:
CANCEL <a class="moz-txt-link-abbreviated" href="mailto:sip:0031*********@sip.xs4all.nl">sip:0031*********@sip.xs4all.nl</a> SIP/2.0
Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK07c2ed92;rport
From: "**********" <a class="moz-txt-link-rfc2396E" href="mailto:sip:*********@sip.xs4all.nl"><sip:*********@sip.xs4all.nl></a>;tag=as70e84199
To: <a class="moz-txt-link-rfc2396E" href="mailto:sip:0031*********@sip.xs4all.nl"><sip:0031*********@sip.xs4all.nl></a>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:740540ee64fa957513ce89f03ef5e3f2@sip.xs4all.nl">740540ee64fa957513ce89f03ef5e3f2@sip.xs4all.nl</a>
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
----------------------------------------------------------------------
However when I dial exactly the same from VoipBuster, I see this instead:
----------------------------------------------------------------------
<--- SIP read from 77.72.169.129:5060 --->
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 195.164.89.135:5060;branch=z9hG4bK6d7efb43;rport
From: "*********" <a class="moz-txt-link-rfc2396E" href="mailto:sip:*********@sip.voipbuster.com"><sip:*********@sip.voipbuster.com></a>;tag=as1374705a
To: <a class="moz-txt-link-rfc2396E" href="mailto:sip:0031*********@sip.voipbuster.com"><sip:0031*********@sip.voipbuster.com></a>;tag=120113ac4a54a269af9e2c
Contact: <a class="moz-txt-link-abbreviated" href="mailto:sip:0031*********@77.72.169.129:5060">sip:0031*********@77.72.169.129:5060</a>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:1949e0303d52a19b1b4f91f16ff94297@sip.voipbuster.com">1949e0303d52a19b1b4f91f16ff94297@sip.voipbuster.com</a>
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 162
v=0
o=********* 1251932194 1251932194 IN IP4 194.221.62.33
s=SIP Call
c=IN IP4 194.221.62.33
t=0 0
m=audio 8958 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
<------------->
--- (11 headers 8 lines) ---
Found RTP audio format 0
Peer audio RTP is at port 194.221.62.33:8958
Found audio description format PCMU for ID 0
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0
(nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing),
combined - 0x0 (nothing)
Peer audio RTP is at port 194.221.62.33:8958
-- SIP/********-089dc538 is making progress passing it to IAX2/2104-8077
== Connect attempt from '127.0.0.1' unable to authenticate
Scheduling destruction of SIP dialog
'<a class="moz-txt-link-abbreviated" href="mailto:1949e0303d52a19b1b4f91f16ff94297@sip.voipbuster.com">1949e0303d52a19b1b4f91f16ff94297@sip.voipbuster.com</a>' in 6400 ms
(Method: INVITE)
Reliably Transmitting (NAT) to 77.72.169.129:5060:
CANCEL <a class="moz-txt-link-abbreviated" href="mailto:sip:0031*********@sip.voipbuster.com">sip:0031*********@sip.voipbuster.com</a> SIP/2.0
Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK6d7efb43;rport
From: "**********" <a class="moz-txt-link-rfc2396E" href="mailto:sip:**********@sip.voipbuster.com"><sip:**********@sip.voipbuster.com></a>;tag=as1374705a
To: <a class="moz-txt-link-rfc2396E" href="mailto:sip:0031*********@sip.voipbuster.com"><sip:0031*********@sip.voipbuster.com></a>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:1949e0303d52a19b1b4f91f16ff94297@sip.voipbuster.com">1949e0303d52a19b1b4f91f16ff94297@sip.voipbuster.com</a>
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
----------------------------------------------------------------------
As you can see, there are different packets being sent, and in the 2nd
case, there is no "is ringing" message, which is rather irritating...
Any suggestions would be appreciated...
TIA
</pre>
</blockquote>
<pre wrap=""><!---->BTW: I am talking about the ringtone the caller should hear... The other
side is ringing, and calls are established just fine, but it is very
irritating to hear nothing until the call either fails or gets picked up...
</pre>
</blockquote>
NM! Found out this only happens on a single extension, and that one was
using IAX... Changed it to SIP as well and got ringing there too!<br>
<br>
-- <br>
FP<br>
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